FIG. 1 generally depicts a digital network 100 adapted to transport voice, data and other information. This exemplary network will be used to illustrate some of the drawbacks associated with present networking communication methodologies and equipment. The network includes a plurality of nodes 102, 104 (hereinafter “network nodes”), each of which may be coupled to various customer premises equipment (CPE) (not shown) and/or other network nodes (not shown). Network nodes 102 and 104 are communicatively coupled to one another through a communication link 106, which may be a wired or wireless communication link. In this network 100, nodes 102 and 104 communicate across communication link 106 using the well-known Asynchronous Transfer Mode (ATM) protocol. Thus, at each end of communication link 106, ATM interfaces 108 are employed. Communication between nodes 102 and 104 with various CPE and/or other nodes is supported across communication links 108 and 110, respectively, each of which is associated with an appropriate interface 112, 114 at its respective node.
When a communication (e.g., a voice communication) is to be transported within network 100, the signal first travels from the associated CPE equipment (e.g., a private branch exchange or PBX) to node 102 over communication link (e.g., digital T1 carrier) 108. Before being transmitted over such a T1 carrier, the signal is sampled and converted to a digital signal. A common sampling rate used with voice communications is 8000 samples per second, with each digital sample represented by 8 bits of data. Thus, the data rate of the new digital signal is: 8000 samples/sec×8 bits=64,000 bits/sec. This technique is known as Pulse Code Modulation (hereinafter “PCM”) and is used extensively throughout the backbone of modern telephone systems.
Although no international standard has been adopted, the T1 carrier is one method of PCM used throughout North America and Japan. The T1 carrier is comprised of 24 channels of digital data multiplexed together. Digitally sampled data from each of the 24 channels are packaged into successive frames of 8 bits/channel×24 channels+an additional framing bit=193 bits. Outside of North America and Japan a similar standard, known as E1, is commonly implemented. E1 operates in a manner similar to T1 except that it uses 32, 8-bit data samples (i.e., 32 channels) instead of 24.
After the signal has been sampled, converted to a digital signal and transmitted over a T1 carrier, as described above, it is then transferred to an outgoing communication link 106. Because the transport protocol across communication link 106 is different than that used on communication link 108, the digital data samples are packaged according to the protocol used across communication link 106 (e.g., ATM) before being transmitted to node 104. Additionally, although PCM by itself provides for a data transfer rate of 64,000 bits/second, it is often desirable to further compress the digital PCM data in order to save bandwidth within the network. This can be accomplished using Digital Signal Processing (hereinafter “DSP”) resources associated with network node 102. For example, if the 64,000 bits/second PCM signal is compressed by a DSP resource at a compression ratio of 16:1, the resulting digital signal will be transmitted at 4,000 bits/second. This represents a significant reduction in required bandwidth across network 100 to transmit the same underlying signal. Such compression techniques are particularly useful in networks that are heavily loaded with network traffic. Examples of compression algorithms known in the art include the International Telegraph Union (hereinafter “ITU’) standards G.711, G.726, G.729-A, G.729, and G.728. Such compression resources may be associated with the ATM interfaces 108 and may operate under the control of a node controller 116 in each of the nodes.
However, there is a tradeoff between bandwidth savings over network 100 and the implementation of costly DSP resources at the nodes. In general, the higher the compression ratio required by the compression algorithm, the more DSP resources are used up processing the compression request over a given period of time. Thus, while a single DSP resource may process up to say 16 channels of data if no compression is used (i.e., in baseline PCM mode), it may be limited to 5 channels if data is compressed at 2:1, and only 2 channels if the PCM signal is compressed at 8:1. One factor behind this limitation is the limited period of time in which the DSP resource must compress the data within a T1 frame before it must move on to the next frame of data. Thus, the chosen compression ratio will have a significant impact on DSP resource usage.
Following compression (if used), the data samples are delivered through network 100 to node 104, where the data may be decompressed and passed on to other CPEs or another node. The system is bi-directional to ensure 2-way communication between the nodes.
One problem with the communication scheme adopted in network 100 occurs when communication link 106 becomes congested, that is, when there is no available bandwidth to support new incoming calls from a CPE coupled to node 102. Consider, for example, a situation where multiple calls being transported between nodes 102 and 104 are using all or almost all of the available bandwidth on communication link 106. If a high priority call (e.g., a 911 or other emergency call) is now received at network node 102, either of two scenarios is possible. First, the high priority call may be rejected (dropped) in the face of no available bandwidth. Second, rather than dropping the high priority call (clearly a least acceptable solution); the nodes may be configured to drop lower priority calls in order to free up bandwidth to accommodate the high priority call. Although this solution may allow the high priority call to proceed, it is less than satisfactory in as much as several existing calls may be dropped to support the one new call. What is needed, therefore, is a more robust mechanism for handling such situations.